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Sunday, September 28, 2008
[asterisk-biz] India Mobile 2.9 Rest of india 3.2
Re: [asterisk-biz] proper analog behavior using an ATA
Steve,
Sorry if my email was packed with TMI. It was not meant to annoy anyone. I just wanted it clear I am not analog stupid and had a solid reference for ATA performance expectations.
> Just for future reference, you don't really compile Asterisk@home, it is just an image that you install.
I said "Started with compiling my own and Asterisk@home." and should have said "started with compiling my own with raw editing of .conf files THEN trying Asterisk@home as a packaged solution".
Thanks for your comments and suggestions.
Jim
----- Original Message -----
From: "Steve Totaro" <stotaro@totarotechnologies.com>
To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk-biz@lists.digium.com>
Sent: Saturday, September 27, 2008 2:40:17 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-biz] proper analog behavior using an ATA
Correction, bibliography = biography
On Sat, Sep 27, 2008 at 3:34 PM, Steve Totaro < stotaro@totarotechnologies.com > wrote:
See comments inline.
On Sat, Sep 27, 2008 at 12:42 PM, Jim Houser < jhouser@trustamerifirst.com > wrote:
Hi all,
Sorry, kinda long but please read...
I'm looking for some help or correction if I'm overlooking something. Let me preface this with I would be "the old guy on the block". I was installing channel banks from Rockwell when they were the size of a fridge for only 48 circuits. Pre-Newbridge days when you had BIG cards for each circuit with dip switches not software. :-)
Why the bibliography? I understand that you have been doing telephony for a long time. So have I, but just not as long as you, same understanding though.
I've dealt with most big name PBXs, Centrex, etc through the years. I have a good data networking background and have a good grasp of common programming languages. I have evolved with the industry, now I'm into VoIP and loving every minute of it. I have been using Asterisk around 3 years. Started with compiling my own and Asterisk@home.
Most "Old School Telephony Guys" Hate VoIP. That may not be the case with you, but most people don't like change in general.
Just for future reference, you don't really compile Asterisk@home, it is just an image that you install.
Here's my issue I hope to get feedback and help with;
I have used many a SIP phone and by way of tweaking * and the phone's local dial plan I have been able to absolutely emulate the behavior and speed of dial out with any TDM system and their priority digital phones. Sound quality has also been matched if not better on the VoIP deployment verses the TDM deployment.
However, this is NOT the case with analog phones. I have used analog FXS adapters from Linksys, Grandstream, Audiocodes and both Digium's analog cards in their TDM400 and the IAXy.
Give Quintum a try, they are excellent. I have heard good things about Rhino and Xorcom. Do you know they stopped using 25 pair lines for stations, that could be part of the problem ;-P
My issues have been proper passing of CID, support for hook flash in small caller id call waiting dependent systems, (home offices and churches), not to mention some installs requiring a bunch of tweaking to kill echo or volume issues. The hook flash support is faulty at times and full of clucks, clicks, slow returning dial tone. Basically a real feeling of cheap quality and "emulation" going on.
In the past, on TDM systems, I used their ATA or a KSU or PBX analog port for any basic analog phone and it was both plug & play along with solid sound quality at all times.
To be fair, it was not plug and play, you had to be somewhat skilled at the switch you were configuring and good with a 110/66 block and a punch too. Many times you needed a couple of people to bust out the toner to make sure you had the right pair.
Heck I even placed modems or faxes behind them without issues, (yes, I understand why a modem or fax is an issue behind the VoIP to FXS conversion). Just throwing this out because it's another area where VoIP is behind the times.
If I were you and I never will be, but I would try to adapt to the paradigm that VoIP is not behind the times, it is just trying to accommodate other technologies that are behind the times, such as modems and FAXs.
Now don't get me wrong. I'm a major Asterisk evangelist and not pushing go back to TDM. My basis for crying for help here is we cannot forget the users of the world were trained and lived on TDM, both in business and at home.
My mother can use a computer very well (and she was "trained" on a typewriter) and my niece is amazing with technnology for her age.
Things get old and antiquated. Morse Code, VCRs, reel to reels, switchboards, betamax, even OTA non digitial TV in a few months. Everything changes, gets better, and people cling to the old ways. I know several people that swear that their old records and record player sound better than a digital CD.
That's where their expectation is. What you sell better sound and work, in it's worst case, like the old TDM platforms did.
And it does, at a MUCH cheaper price point, with many more features, if engineered and installed by someone who knows what they are doing. Flexibility is mind blowing.
I should mention I have obtained the level and quality in an analog phone that is top notch without the emulation feel but it is only for the large users. That has been to do a Asterisk T1 connected to an Audit 600 using analog station cards. This paralleled the analog service delivery I could get from the TDM world, but it's an expensive deployment.
I think you should try a few different vendors and solutions. What you mention above is going to give you pretty much the best analog. I am sure someone else can sell you something that does this with SIP. My personal recommendation is Quintum, but I am sure others do just as well.
What have people used in a small deployment, 2 to 4 FXS ports, that REALLY performs like a traditional TDM delivered analog service?
Generally, you buy a Digium or Sangoma board with the right number of FXO ports and then in install Polycom (or whatever) SIP phones.
Personally, with both Digium and Sangoma FXS ports, I get perfect operation. Maybe try posting your configs instead of a book and biography.
Have you explored your zap configs? There could be a simple setting that makes everything wonderful. Asterisk@home didn't really have much facility for that.
Thanks for allowing me to rant, (fighting with a Linksys PAP2T on a home office Asterisk switch right now).
Jim
Thanks,
Steve Totaro
_______________________________________________
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Re: [asterisk-biz] proper analog behavior using an ATA
Please explain the difference between an anaolg card on a digital System that you have such high regard for? Is it just simply the lack of DSPs on the cards, the quality, what specifically?
They do the same thing in function.
I heard nothing but complaining but not a bit of room to help you. Anyways, selling Channel banks should not be a problem since it sounds like you sell into the pretty high end market.
Thanks,
Steve Totaro
On Sun, Sep 28, 2008 at 5:50 PM, Jim Houser <jhouser@trustamerifirst.com> wrote:
Alex,
Please let your friend know I said "thanks" for his comments. I agree with his thinking.
Jim
----- Original Message -----
From: "Alex Balashov" <abalashov@evaristesys.com>
To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk-biz@lists.digium.com>
Sent: Sunday, September 28, 2008 4:42:11 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-biz] proper analog behavior using an ATA
This opinion is offered by a friend of mine who saw the message, but who
is not on the list. Credit goes to Sean McCord (scm!!at!!cycoresys.com).
---
"The _only_ way to do analog right is with dedicated analog
hardware: a channel bank. You can "get it working" with ATAs and PCI
cards, but you will _always_ be fighting problems. There will be plenty
of proponents of ATA and Digium cards for analog saying that they work
just fine. There always are. For a production system in which you will
actually be using features on the phone beyond the basic "connect a call
to the PSTN," and, more importantly, interfacing with other analog gear,
you will only drive yourself crazy and waste more money in time and
effort getting them to work.
Inevitably, the voltages are wrong, the ringing is insufficient, the
transmission levels are incorrect, the signalling isn't reliably passed,
timings are not what they should be, etc. IMHO, these devices are made
by people and companies which are VoIP first and analog
"Johnny-come-lately"'s. I have fought too many battles and have eaten
the purchase of too many channel banks when all else failed to ever
properly recommend ATA for analog PCI cards for most any professional
installation.
This is certainly true of Sipura/Linksys, Grandstream, Digium, and
Sangoma, by my own experience. I am tentatively hopeful of the Rhino
analog PCI cards, but I only have one of those in the field on a system
which has light traffic and limited need of features.
Note that this distrust does not extend to a significant extent to
digital PCI cards. I have generally not had nearly the problems (after
the first or second generation Digium cards, anyway) with them. There
_are_ problems, of course, but they tend to be "hard" problems, not the
intermittant, degrading-over-time types of problems as with the analog
gear, so they can, at least, be dealt with.
Channel bank-based solutions _are_ more expensive on the hardware, and I
don't really see a way around that, but the trade-off is that, over
time, they will be _much_ _less_ expensive due to the lower maintenance
and troubleshooting costs. I know it's a hard sell, but it is
definitely worth it.
As for alternatives to Adits, the CAC AccessBanks (I or II) can be
obtained cheaply on eBay and the like, if you get lucky (though watch
that the ABIs are not TR-08s, which will not work). Rhino also makes
excellent channel banks, and if you are going to buy new equipment (as
opposed to used or eBay eq), they offer a good price, as well."
---
Jim Houser wrote:
> Hi all,
>
> Sorry, kinda long but please read...
>
> I'm looking for some help or correction if I'm overlooking something. Let me preface this with I would be "the old guy on the block". I was installing channel banks from Rockwell when they were the size of a fridge for only 48 circuits. Pre-Newbridge days when you had BIG cards for each circuit with dip switches not software. :-)
>
> I've dealt with most big name PBXs, Centrex, etc through the years. I have a good data networking background and have a good grasp of common programming languages. I have evolved with the industry, now I'm into VoIP and loving every minute of it. I have been using Asterisk around 3 years. Started with compiling my own and Asterisk@home.
>
> Here's my issue I hope to get feedback and help with;
>
> I have used many a SIP phone and by way of tweaking * and the phone's local dial plan I have been able to absolutely emulate the behavior and speed of dial out with any TDM system and their priority digital phones. Sound quality has also been matched if not better on the VoIP deployment verses the TDM deployment.
>
> However, this is NOT the case with analog phones. I have used analog FXS adapters from Linksys, Grandstream, Audiocodes and both Digium's analog cards in their TDM400 and the IAXy.
>
> My issues have been proper passing of CID, support for hook flash in small caller id call waiting dependent systems, (home offices and churches), not to mention some installs requiring a bunch of tweaking to kill echo or volume issues. The hook flash support is faulty at times and full of clucks, clicks, slow returning dial tone. Basically a real feeling of cheap quality and "emulation" going on.
>
> In the past, on TDM systems, I used their ATA or a KSU or PBX analog port for any basic analog phone and it was both plug & play along with solid sound quality at all times. Heck I even placed modems or faxes behind them without issues, (yes, I understand why a modem or fax is an issue behind the VoIP to FXS conversion). Just throwing this out because it's another area where VoIP is behind the times.
>
> Now don't get me wrong. I'm a major Asterisk evangelist and not pushing go back to TDM. My basis for crying for help here is we cannot forget the users of the world were trained and lived on TDM, both in business and at home. That's where their expectation is. What you sell better sound and work, in it's worst case, like the old TDM platforms did.
>
> I should mention I have obtained the level and quality in an analog phone that is top notch without the emulation feel but it is only for the large users. That has been to do a Asterisk T1 connected to an Audit 600 using analog station cards. This paralleled the analog service delivery I could get from the TDM world, but it's an expensive deployment.
>
> What have people used in a small deployment, 2 to 4 FXS ports, that REALLY performs like a traditional TDM delivered analog service?
>
> Thanks for allowing me to rant, (fighting with a Linksys PAP2T on a home office Asterisk switch right now).
>
> Jim
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-biz
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
_______________________________________________
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
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--
Thanks,
Steve Totaro
1.888.777.1888
1.240.938.1212 (cell)
Re: [asterisk-biz] proper analog behavior using an ATA
Alex,
Please let your friend know I said "thanks" for his comments. I agree with his thinking.
Jim
----- Original Message -----
From: "Alex Balashov" <abalashov@evaristesys.com>
To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk-biz@lists.digium.com>
Sent: Sunday, September 28, 2008 4:42:11 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-biz] proper analog behavior using an ATA
This opinion is offered by a friend of mine who saw the message, but who
is not on the list. Credit goes to Sean McCord (scm!!at!!cycoresys.com).
---
"The _only_ way to do analog right is with dedicated analog
hardware: a channel bank. You can "get it working" with ATAs and PCI
cards, but you will _always_ be fighting problems. There will be plenty
of proponents of ATA and Digium cards for analog saying that they work
just fine. There always are. For a production system in which you will
actually be using features on the phone beyond the basic "connect a call
to the PSTN," and, more importantly, interfacing with other analog gear,
you will only drive yourself crazy and waste more money in time and
effort getting them to work.
Inevitably, the voltages are wrong, the ringing is insufficient, the
transmission levels are incorrect, the signalling isn't reliably passed,
timings are not what they should be, etc. IMHO, these devices are made
by people and companies which are VoIP first and analog
"Johnny-come-lately"'s. I have fought too many battles and have eaten
the purchase of too many channel banks when all else failed to ever
properly recommend ATA for analog PCI cards for most any professional
installation.
This is certainly true of Sipura/Linksys, Grandstream, Digium, and
Sangoma, by my own experience. I am tentatively hopeful of the Rhino
analog PCI cards, but I only have one of those in the field on a system
which has light traffic and limited need of features.
Note that this distrust does not extend to a significant extent to
digital PCI cards. I have generally not had nearly the problems (after
the first or second generation Digium cards, anyway) with them. There
_are_ problems, of course, but they tend to be "hard" problems, not the
intermittant, degrading-over-time types of problems as with the analog
gear, so they can, at least, be dealt with.
Channel bank-based solutions _are_ more expensive on the hardware, and I
don't really see a way around that, but the trade-off is that, over
time, they will be _much_ _less_ expensive due to the lower maintenance
and troubleshooting costs. I know it's a hard sell, but it is
definitely worth it.
As for alternatives to Adits, the CAC AccessBanks (I or II) can be
obtained cheaply on eBay and the like, if you get lucky (though watch
that the ABIs are not TR-08s, which will not work). Rhino also makes
excellent channel banks, and if you are going to buy new equipment (as
opposed to used or eBay eq), they offer a good price, as well."
---
Jim Houser wrote:
> Hi all,
>
> Sorry, kinda long but please read...
>
> I'm looking for some help or correction if I'm overlooking something. Let me preface this with I would be "the old guy on the block". I was installing channel banks from Rockwell when they were the size of a fridge for only 48 circuits. Pre-Newbridge days when you had BIG cards for each circuit with dip switches not software. :-)
>
> I've dealt with most big name PBXs, Centrex, etc through the years. I have a good data networking background and have a good grasp of common programming languages. I have evolved with the industry, now I'm into VoIP and loving every minute of it. I have been using Asterisk around 3 years. Started with compiling my own and Asterisk@home.
>
> Here's my issue I hope to get feedback and help with;
>
> I have used many a SIP phone and by way of tweaking * and the phone's local dial plan I have been able to absolutely emulate the behavior and speed of dial out with any TDM system and their priority digital phones. Sound quality has also been matched if not better on the VoIP deployment verses the TDM deployment.
>
> However, this is NOT the case with analog phones. I have used analog FXS adapters from Linksys, Grandstream, Audiocodes and both Digium's analog cards in their TDM400 and the IAXy.
>
> My issues have been proper passing of CID, support for hook flash in small caller id call waiting dependent systems, (home offices and churches), not to mention some installs requiring a bunch of tweaking to kill echo or volume issues. The hook flash support is faulty at times and full of clucks, clicks, slow returning dial tone. Basically a real feeling of cheap quality and "emulation" going on.
>
> In the past, on TDM systems, I used their ATA or a KSU or PBX analog port for any basic analog phone and it was both plug & play along with solid sound quality at all times. Heck I even placed modems or faxes behind them without issues, (yes, I understand why a modem or fax is an issue behind the VoIP to FXS conversion). Just throwing this out because it's another area where VoIP is behind the times.
>
> Now don't get me wrong. I'm a major Asterisk evangelist and not pushing go back to TDM. My basis for crying for help here is we cannot forget the users of the world were trained and lived on TDM, both in business and at home. That's where their expectation is. What you sell better sound and work, in it's worst case, like the old TDM platforms did.
>
> I should mention I have obtained the level and quality in an analog phone that is top notch without the emulation feel but it is only for the large users. That has been to do a Asterisk T1 connected to an Audit 600 using analog station cards. This paralleled the analog service delivery I could get from the TDM world, but it's an expensive deployment.
>
> What have people used in a small deployment, 2 to 4 FXS ports, that REALLY performs like a traditional TDM delivered analog service?
>
> Thanks for allowing me to rant, (fighting with a Linksys PAP2T on a home office Asterisk switch right now).
>
> Jim
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-biz
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-biz] proper analog behavior using an ATA
This opinion is offered by a friend of mine who saw the message, but who
is not on the list. Credit goes to Sean McCord (scm!!at!!cycoresys.com).
is not on the list. Credit goes to Sean McCord (scm!!at!!cycoresys.com).
---
"The _only_ way to do analog right is with dedicated analog
hardware: a channel bank. You can "get it working" with ATAs and PCI
cards, but you will _always_ be fighting problems. There will be plenty
of proponents of ATA and Digium cards for analog saying that they work
just fine. There always are. For a production system in which you will
actually be using features on the phone beyond the basic "connect a call
to the PSTN," and, more importantly, interfacing with other analog gear,
you will only drive yourself crazy and waste more money in time and
effort getting them to work.
Inevitably, the voltages are wrong, the ringing is insufficient, the
transmission levels are incorrect, the signalling isn't reliably passed,
timings are not what they should be, etc. IMHO, these devices are made
by people and companies which are VoIP first and analog
"Johnny-come-lately"'s. I have fought too many battles and have eaten
the purchase of too many channel banks when all else failed to ever
properly recommend ATA for analog PCI cards for most any professional
installation.
This is certainly true of Sipura/Linksys, Grandstream, Digium, and
Sangoma, by my own experience. I am tentatively hopeful of the Rhino
analog PCI cards, but I only have one of those in the field on a system
which has light traffic and limited need of features.
Note that this distrust does not extend to a significant extent to
digital PCI cards. I have generally not had nearly the problems (after
the first or second generation Digium cards, anyway) with them. There
_are_ problems, of course, but they tend to be "hard" problems, not the
intermittant, degrading-over-time types of problems as with the analog
gear, so they can, at least, be dealt with.
Channel bank-based solutions _are_ more expensive on the hardware, and I
don't really see a way around that, but the trade-off is that, over
time, they will be _much_ _less_ expensive due to the lower maintenance
and troubleshooting costs. I know it's a hard sell, but it is
definitely worth it.
As for alternatives to Adits, the CAC AccessBanks (I or II) can be
obtained cheaply on eBay and the like, if you get lucky (though watch
that the ABIs are not TR-08s, which will not work). Rhino also makes
excellent channel banks, and if you are going to buy new equipment (as
opposed to used or eBay eq), they offer a good price, as well."
---
Jim Houser wrote:
> Hi all,
>
> Sorry, kinda long but please read...
>
> I'm looking for some help or correction if I'm overlooking something. Let me preface this with I would be "the old guy on the block". I was installing channel banks from Rockwell when they were the size of a fridge for only 48 circuits. Pre-Newbridge days when you had BIG cards for each circuit with dip switches not software. :-)
>
> I've dealt with most big name PBXs, Centrex, etc through the years. I have a good data networking background and have a good grasp of common programming languages. I have evolved with the industry, now I'm into VoIP and loving every minute of it. I have been using Asterisk around 3 years. Started with compiling my own and Asterisk@home.
>
> Here's my issue I hope to get feedback and help with;
>
> I have used many a SIP phone and by way of tweaking * and the phone's local dial plan I have been able to absolutely emulate the behavior and speed of dial out with any TDM system and their priority digital phones. Sound quality has also been matched if not better on the VoIP deployment verses the TDM deployment.
>
> However, this is NOT the case with analog phones. I have used analog FXS adapters from Linksys, Grandstream, Audiocodes and both Digium's analog cards in their TDM400 and the IAXy.
>
> My issues have been proper passing of CID, support for hook flash in small caller id call waiting dependent systems, (home offices and churches), not to mention some installs requiring a bunch of tweaking to kill echo or volume issues. The hook flash support is faulty at times and full of clucks, clicks, slow returning dial tone. Basically a real feeling of cheap quality and "emulation" going on.
>
> In the past, on TDM systems, I used their ATA or a KSU or PBX analog port for any basic analog phone and it was both plug & play along with solid sound quality at all times. Heck I even placed modems or faxes behind them without issues, (yes, I understand why a modem or fax is an issue behind the VoIP to FXS conversion). Just throwing this out because it's another area where VoIP is behind the times.
>
> Now don't get me wrong. I'm a major Asterisk evangelist and not pushing go back to TDM. My basis for crying for help here is we cannot forget the users of the world were trained and lived on TDM, both in business and at home. That's where their expectation is. What you sell better sound and work, in it's worst case, like the old TDM platforms did.
>
> I should mention I have obtained the level and quality in an analog phone that is top notch without the emulation feel but it is only for the large users. That has been to do a Asterisk T1 connected to an Audit 600 using analog station cards. This paralleled the analog service delivery I could get from the TDM world, but it's an expensive deployment.
>
> What have people used in a small deployment, 2 to 4 FXS ports, that REALLY performs like a traditional TDM delivered analog service?
>
> Thanks for allowing me to rant, (fighting with a Linksys PAP2T on a home office Asterisk switch right now).
>
> Jim
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-biz
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Telephony Depot is hiring
Folks,
Telephony Depot is always looking for talented people who get excited
about open telephony platforms such as Asterisk (in its various
forms), FreeSwitch, Yate etc, and who just plain find telephony
hardware strangely fascinating ;-) Recently though, we've decided to
add more sales, support and programming depth to our bench and have
immediate need for a few people in our Philadelphia office. We're
looking primarily for:
- inside sales rep
- entry-level tech support engineer
- web developer / programmer
but we're a young & growing company (Inc. 500 two years running now)
and we're definitely open to any other strategic hire that might help
take us to the next level of growth. If you think you have something
to contribute to online merchandising of hardware around open
telephony, send your resume and a brief introduction to work@our-
domain-name (sorry email harvesters!). Telecommuting may be an option
for some positions, so if Philadelphia is a non-starter please don't
assume this rules you out!
Sincerely,
--
Darren Nickerson
Telephony Depot
www.telephonydepot.com
+1.215.825.8710 ext 8106 (office)
+1.215.243.8335 (fax)
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Saturday, September 27, 2008
Re: [asterisk-biz] proper analog behavior using an ATA
Correction, bibliography = biography
On Sat, Sep 27, 2008 at 3:34 PM, Steve Totaro <stotaro@totarotechnologies.com> wrote:
See comments inline.On Sat, Sep 27, 2008 at 12:42 PM, Jim Houser <jhouser@trustamerifirst.com> wrote:
Hi all,
Sorry, kinda long but please read...
I'm looking for some help or correction if I'm overlooking something. Let me preface this with I would be "the old guy on the block". I was installing channel banks from Rockwell when they were the size of a fridge for only 48 circuits. Pre-Newbridge days when you had BIG cards for each circuit with dip switches not software. :-)
Why the bibliography? I understand that you have been doing telephony for a long time. So have I, but just not as long as you, same understanding though.
I've dealt with most big name PBXs, Centrex, etc through the years. I have a good data networking background and have a good grasp of common programming languages. I have evolved with the industry, now I'm into VoIP and loving every minute of it. I have been using Asterisk around 3 years. Started with compiling my own and Asterisk@home.
Most "Old School Telephony Guys" Hate VoIP. That may not be the case with you, but most people don't like change in general.
Just for future reference, you don't really compile Asterisk@home, it is just an image that you install.
Here's my issue I hope to get feedback and help with;
I have used many a SIP phone and by way of tweaking * and the phone's local dial plan I have been able to absolutely emulate the behavior and speed of dial out with any TDM system and their priority digital phones. Sound quality has also been matched if not better on the VoIP deployment verses the TDM deployment.
However, this is NOT the case with analog phones. I have used analog FXS adapters from Linksys, Grandstream, Audiocodes and both Digium's analog cards in their TDM400 and the IAXy.
Give Quintum a try, they are excellent. I have heard good things about Rhino and Xorcom. Do you know they stopped using 25 pair lines for stations, that could be part of the problem ;-P
My issues have been proper passing of CID, support for hook flash in small caller id call waiting dependent systems, (home offices and churches), not to mention some installs requiring a bunch of tweaking to kill echo or volume issues. The hook flash support is faulty at times and full of clucks, clicks, slow returning dial tone. Basically a real feeling of cheap quality and "emulation" going on.
In the past, on TDM systems, I used their ATA or a KSU or PBX analog port for any basic analog phone and it was both plug & play along with solid sound quality at all times.
To be fair, it was not plug and play, you had to be somewhat skilled at the switch you were configuring and good with a 110/66 block and a punch too. Many times you needed a couple of people to bust out the toner to make sure you had the right pair.
Heck I even placed modems or faxes behind them without issues, (yes, I understand why a modem or fax is an issue behind the VoIP to FXS conversion). Just throwing this out because it's another area where VoIP is behind the times.
If I were you and I never will be, but I would try to adapt to the paradigm that VoIP is not behind the times, it is just trying to accommodate other technologies that are behind the times, such as modems and FAXs.
My mother can use a computer very well (and she was "trained" on a typewriter) and my niece is amazing with technnology for her age.
Now don't get me wrong. I'm a major Asterisk evangelist and not pushing go back to TDM. My basis for crying for help here is we cannot forget the users of the world were trained and lived on TDM, both in business and at home.
Things get old and antiquated. Morse Code, VCRs, reel to reels, switchboards, betamax, even OTA non digitial TV in a few months. Everything changes, gets better, and people cling to the old ways. I know several people that swear that their old records and record player sound better than a digital CD.And it does, at a MUCH cheaper price point, with many more features, if engineered and installed by someone who knows what they are doing. Flexibility is mind blowing.That's where their expectation is. What you sell better sound and work, in it's worst case, like the old TDM platforms did.
I should mention I have obtained the level and quality in an analog phone that is top notch without the emulation feel but it is only for the large users. That has been to do a Asterisk T1 connected to an Audit 600 using analog station cards. This paralleled the analog service delivery I could get from the TDM world, but it's an expensive deployment.
I think you should try a few different vendors and solutions. What you mention above is going to give you pretty much the best analog. I am sure someone else can sell you something that does this with SIP. My personal recommendation is Quintum, but I am sure others do just as well.
What have people used in a small deployment, 2 to 4 FXS ports, that REALLY performs like a traditional TDM delivered analog service?
Generally, you buy a Digium or Sangoma board with the right number of FXO ports and then in install Polycom (or whatever) SIP phones.
Personally, with both Digium and Sangoma FXS ports, I get perfect operation. Maybe try posting your configs instead of a book and biography.
Have you explored your zap configs? There could be a simple setting that makes everything wonderful. Asterisk@home didn't really have much facility for that.
Thanks for allowing me to rant, (fighting with a Linksys PAP2T on a home office Asterisk switch right now).
Jim
Thanks,
Steve Totaro
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